Rate control is essential for media streaming over packet networks. The challenge in delivering bandwidth-intensive content like multimedia over capacity-limited, shared links is to quickly respond to changes in network conditions by adjusting the bitrate and the media encoding scheme to optimize the viewing and listening experience of the user. In particular, when transferring a fixed bitrate over a connection that cannot provide the necessary throughput, several undesirable effects arise. For example, a network buffer may overflow resulting in packet loss causing garbled video or audio playback, or a media player buffer may underflow resulting in playback stall. Standard bodies have recommended protocols to address these issues. Internet Engineering Task Force (IETF), in RFC 3550, specifies RTCP as the fundamental building block to implement bit rate/packet rate control in streaming media. Several extensions to RTCP, suited for high capacity networks, follow this original recommendation.
Even with these recommended protocols, delivering a multimedia session over wireless networks can be particularly challenging, due in part to the following:                Sudden Adjustment of nominal transmission rate: Due to interference, fading, etc, 3+G networks negotiate physical layer parameters on the fly. Nominal transmission bitrates can change by a factor of 10;        Packet Loss: caused by either link transmission errors or by network congestion;        Reduction of Effective bandwidth: The wireless link is a shared resource at Layer 2, with MAC (Media Access Control) mechanism and scheduling. This means that an increased load presented by other wireless terminals in the same sector can reduce the effective bandwidth or capacity that a terminal will see; and        Limited Capacity: Available capacity is typically a fraction to that obtained in traditional wireline internet access technologies, where currently capacity is not an issue. Fixed internet media sessions can typically offer to the network loads between 250 and 400 kbps. Despite the fact that current 3G cellular networks can sustain throughputs of 500 kbps and above, the total bitrate budget for a wireless multimedia session is typically kept under 150 kbps to ensure scalability.        
For wireless mobile devices, providing a good experience in streaming media sessions is particularly difficult, due to                Infrequent and incomplete network state information. The typical wireless media player support RTCP receiver report as defined in RFC 3550, and the report generation frequency is fixed. As a result, the network state information obtained at the sender end is limited and sporadic. In its Packet Streaming Service specification, 3GPP recommends several extensions to the basic IETF RTCP Receiver Report (i.e. RTCP Extended Reports, or XR). Unfortunately, very few handsets implement these enhancements;        Different media streams are handled separately. Despite the fact that they are both transmitted over the same network link, audio and video streams are handled separately by RTCP. Both RTCP reports provide state information about the same network, therefore a joint analysis; and        Low bitrates available: The bitrate budget for a wireless multimedia session is generally very low (under 150 kbps). The adjustment of audio and video bitrates can have large perceptual impact on the session, and the total available network bitrate, even for 3G networks, can fall well below acceptable quantities. With these issues, to wireless networks and wireless mobile devices it has been difficult to set up a consistent streaming media session.        